Vassilis Konstantinou
2004-Jun-27 02:09 UTC
[Asterisk-Users] Why? oh why can't I dial out?
I have been struggling with my Asterisk setup for 3 days now and I think I have done well...apart from the small detail that I cannot dial out on my phone (PSTN) line. My setup is: Suse Linux 9.0 1 fxo card connected to a BT(UK) line 1 Cisco ATA186 sip v3.0 with two analogue phones attached to it Asterix CVS-HEAD-05/30/04-06:56:31 with the UK Userid patch applied. Asterisk loads without any warnings. The problem? I can receive calls with the userid reported correctly. I can forward them to the two SIP ATA lines. I can dial internally (between the two phones) BUT I cannot dial out :-( I have tried everything (and yes I searched the world using google but nothing seems to apply to my case). So can somebody please direct me to possible causes. The scenario is: if I dial 9123 (for the UK clock) then output from the console is: -- Executing Dial("SIP/5000-96f1", "Zap/1/123") in new stack -- Called 1/123 -- Zap/1-1 answered SIP/5000-96f1 -- Hungup 'Zap/1-1' SIP/5000 is one of my ATA phones ZAP/1 is the fxo card The call is transferred to Zap/1 as I can hear the dial tone but then nothing happens (it does not dial 123). It just stays on the tone until it times out. I also tried pressing buttons on my ATA phone but nothing is transferred. HELP! Here is a collection of my conf files: zaptel.conf fxsks=1 loadzone=uk defaultzone=uk --------------------------- zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; ;context=default context=incoming switchtype=national signalling=fxs_ks usedistinctiveringdetection=no rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=no transfer=no ancallforward=no callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=4.0 immediate=no musiconhold=default busydetect=no callprogress=no usecallerid=uk channel => 1 -------------------------------- part of extensions.conf [incoming] exten => s,1,SetCallerId(${CALLERID}) exten => s,2,dial(SIP/5000&SIP/5001,10,tr) exten => s,3,Voicemail,u1000 exten => s,102,Voicemail,b1000 exten => _9.,1,Dial(${CONSOLE}/${EXTEN:1}) exten => _9.,2,Congestion Vassilis
On Sun, 27 Jun 2004, Vassilis Konstantinou wrote:> I have been struggling with my Asterisk setup for 3 days now and I think I > have done well...apart from the small detail that I cannot dial out on my > phone (PSTN) line. > [snip] > The scenario is: if I dial 9123 (for the UK clock) then output from the > console is: > > -- Executing Dial("SIP/5000-96f1", "Zap/1/123") in new stack > -- Called 1/123 > -- Zap/1-1 answered SIP/5000-96f1 > -- Hungup 'Zap/1-1' >[snip] > > [incoming] > > exten => s,1,SetCallerId(${CALLERID}) > exten => s,2,dial(SIP/5000&SIP/5001,10,tr) > exten => s,3,Voicemail,u1000 > exten => s,102,Voicemail,b1000 > > > exten => _9.,1,Dial(${CONSOLE}/${EXTEN:1}) > exten => _9.,2,CongestionSo, assuming that calls from your SIP device are in the same context as the above extensions, all extensions beginning with a 9 should be dialled on ${CONSOLE}. On my box, ${CONSOLE}=console/dsp... the sound card. Is yours set to something similar (or is it really set to dial the zap interface?) Not being from the UK myself, I don't know whether the clock's number is 123 or 9123. If it's 9123, then you should be dialing 99123 in order to get through your dialplan with the 9123 still intact to send to the PSTN. Greg